Paul Frindle - Product Design
Paul Frindle has 35 years' experience in the pro audio and music industries. He has worked as a studio engineer in Oxford and Paris, and was a design engineer at SSL with responsibilities for E and G-series analogue consoles, emerging assignable consoles and nascent digital audio products. As one of the original team that became Sony Oxford, he is responsible for many revolutionary aspects of the Sony OXF-R3 mixing console. More recently he was responsible for product design and quality assurance at Oxford Plugins. On leaving Sony Oxford, he co-founded Pro Audio DSP in order to make novel sound-processing applications to fulfill many issues he had identified in the audio production chain over his career.
Paul is a very trusted Pro Audio Digital Myth buster here at Gearslutz. He has debated and answered the best questions and has giving a wealth of knowledge throughout this fine forum but, all the info is scattered and he has had to answer the same question over and over and over again.
Well now, hopefully this can be a sticky and a link for My Frindle to revert people back too instead of, writing the same thing over and over again.

So Paul, I guess I will just say that, this is an area for you to take your time, answer in great detail if you would like so that you don't have to re-explain yourself in other threads and just lead others to your "in-detail answer" here….

I'll start off the most requested questions that I see on the internet now a days, and all I ask from Gearsluters is,

Please ask your questions like this…

"Is This Truth Or Myth? -"

That way, he can give an easy answer at first, and explain it for the hardcore digital geeks and pro audio heads here….

PLEASE - NO DEBATING!!!!

If you want to debate with him on something he has answered here, please feel free to start a thread about that issue and PM him to let him know where it's located.
Reason being, this will keep this thread clean and informational for people that just want to know his opinion on that matter and not everyone else's.

Thanx in advanced to all that help this thread become a great "THREAD OF KNOWLEDGE"

So, I'll start it off…..

1. Is This Truth Or Myth? -
Digital can achieve and/or surpass the best analog headroom.

2. Is This Truth Or Myth? -
Digital can truly and fully emulate analog mixing boards of yester year (SSL/NEVE etc).

3. Is This Truth Or Myth? -
Digital system's hardware only achieve 20hz - 20khz, there is no reason to have a plugin @ 10hz - 40khz? We can't hear and/or feel the difference.

4. Is This Truth Or Myth? -
Current metering plugins/systems are totally accurate? Which one would you recommend by the way.

5. Is This Truth Or Myth? -
64bit will help the sound of older and/or current plugins.

6. Is This Truth Or Myth? -
Pro Tools 48bit Mix engine is enough to handle a 192 channel mix accurately? It would not have to be a 56bit, 64bit or even a 72bit mixer.

7. Is This Truth Or Myth? -
In Pro Tools, as you know, On a TDM system, plugins handle like this
TDM = 24bit in to (If plugin is double precision) 48bit to 24 bit out
RTAS = 32bit in to (If plugin is double precision) 64bit to 32bit out
TDM to RTAS to TDM = 24bit in to 48bit to 24bit out, to 32bit in to 64bit to 32bit out, to 24bit in to 48bit to 24bit out.
So it is not good to use TDM to RTAS or RTAS to TDM as too many calculations will only make sound worst and/or Pro Tools mix engine can not handle
all these calculations and will only make your mix/recording worst.

8. Is This Truth Or Myth? -
It is better to stay totally ITB then to use a summing mixer for width/depth/headroom. Minus color of course.

9. Is This Truth Or Myth? -
Even with today's cpu's, there is no way that we could exceed the sound of old reverb units. (Lexicon 480L/PCM90 etc)

10. Is This Truth Or Myth? -
64bit floating point is better than 48bit fixed point. In fact, we should go 72bit to make it more analogy.

Last edited by M2E; 23rd October 2010 at 07:41 PM..
Reason: Looks better and more define

there is no reason to have a plugin @ 10hz - 40khz? We can't hear and/or feel the difference.

Myth.

Quote:

Originally Posted by M2E

5. Is This Truth Or Myth? -
64bit will help the sound of older and/or current plugins.

64 bit what, where ? The plugin sounds like it sounds,a 64 bit OS or host won't change that. Some may only use 32 bit float internally but could benefit from double precision but i suspect that is fairly rare.

Quote:

Originally Posted by M2E

6. Is This Truth Or Myth? -
Pro Tools 48bit Mix engine is enough to handle a 192 channel mix accurately? It would not have to be a 56bit, 64bit or even a 72bit mixer.

Of course it can.

Quote:

Originally Posted by M2E

7. Is This Truth Or Myth? -
In Pro Tools, as you know, On a TDM system, plugins handle like this
TDM = 24bit in to (If plugin is double precision) 48bit to 24 bit out
RTAS = 32bit in to (If plugin is double precision) 64bit to 32bit out
TDM to RTAS to TDM = 24bit in to 48bit to 24bit out, to 32bit in to 64bit to 32bit out, to 24bit in to 48bit to 24bit out.
So it is not good to use TDM to RTAS or RTAS to TDM as too many calculations will only make sound worst and/or Pro Tools mix engine can not handle
all these calculations and will only make your mix/recording worst.

Nonsense.

Quote:

Originally Posted by M2E

8. Is This Truth Or Myth? -
It is better to stay totally ITB then to use a summing mixer for width/depth/headroom. Minus color of course.

That would be subjective,no ?

Quote:

Originally Posted by M2E

9. Is This Truth Or Myth? -
Even with today's cpu's, there is no way that we could exceed the sound of old reverb units. (Lexicon 480L/PCM90 etc)

Complete nonsense.

Quote:

Originally Posted by M2E

10. Is This Truth Or Myth? -64bit floating point is better than 48bit fixed point.

I will do short answers to these for starters, all I will say is my opinion only. As ever - longer answers are only needed where people disagree and require detailed discussion.

Quote:

Originally Posted by M2E

So, I'll start it off…..

1. Is This Truth Or Myth? -
Digital can achieve and/or surpass the best analog headroom.

Truth - because the dynamic range of digital is limited only by mathematical precision, whereas analogue is limited by the physical world which cannot be avoided.

A 24bit signal path with 144dB of dynamic range is just about as good as any analogue process could practically produce.

To provide headroom simply modulate at lower levels to allow overshoots to not be clipped off in fixed point systems and output media.

Quote:

2. Is This Truth Or Myth? -
Digital can truly and fully emulate analog mixing boards of yester year (SSL/NEVE etc).

Truth - if someone went to the trouble of doing so.

Providing the same result is produced for all signals it does not matter how the processing is done. There is no magic.

The biggest single difference between analogue systems of old and the current digital environment is that analogue systems had headroom above the modulation levels - to accomodate over signals and progressive analogue degradation at over normal levels.

A properly implemented digital system does not have this as it is 'correct' at all levels.

Ths changes the workflow significantly and unfortunately encourages people to modulte at maximum, thereby removing degrees of freedom from the work flow.

Quote:

3. Is This Truth Or Myth? -
Digital system's hardware only achieve 20hz - 20khz, there is no reason to have a plugin @ 10hz - 40khz? We can't hear and/or feel the difference.

Truth - if everything is done correcly.

Digital hardware even at base sampling rate can provide 0Hz - 20KHz.

I personally have no evidence (however much I've tried) that anyone can hear directly beyond that range.

Therefore it seems there is nothing much to be gained from producing supersonic freqs we can't hear that serve only to cause errors in repro systems.

However being able to set a centre freq of above 20KHz in something like an EQ still has merit because the tail of the response below that can still be heard and appreciated. But this can be done without sampling the systen at higher rates.

Quote:

4. Is This Truth Or Myth? -
Current metering plugins/systems are totally accurate? Which one would you recommend by the way.

Myth - unfortunately.

Current digital metering systems in workstations measure sample value only. Since this is not decoded signal the meters do not show actual signal level.

On reconstruction (in a DAC for instance) combinations of high level sample values that do not cause overs on the workstation metering can produce signal levels up to +5 or even +6dB above full modulation when decoded. How these get treated in the users equipment is not specified or precise - so errors can be generated that sound bad.

This is commonly called 'inter-sample peaking'. But really it is simply the result of increasing sample values beyond what could be generated by converting signal from the real world, and they are in error of strict sampling theory.

Quote:

5. Is This Truth Or Myth? -
64bit will help the sound of older and/or current plugins.

Myth.

If you mean the latest '64 bit' operating systems, the answer is no.

It refers only to the ability to access greater than 4GB of RAM and has no bearing on mathematical precision, which has not changed.

Quote:

6. Is This Truth Or Myth? -
Pro Tools 48bit Mix engine is enough to handle a 192 channel mix accurately? It would not have to be a 56bit, 64bit or even a 72bit mixer.

Truth. In fact the 48bit processor has a 56bit acumulator so that accuracy is not lost when signals become large in value. There is no problem with the mixer provided it's presented with 24bit signals.

Quote:

7. Is This Truth Or Myth? -
In Pro Tools, as you know, On a TDM system, plugins handle like this
TDM = 24bit in to (If plugin is double precision) 48bit to 24 bit out
RTAS = 32bit in to (If plugin is double precision) 64bit to 32bit out
TDM to RTAS to TDM = 24bit in to 48bit to 24bit out, to 32bit in to 64bit to 32bit out, to 24bit in to 48bit to 24bit out.
So it is not good to use TDM to RTAS or RTAS to TDM as too many calculations will only make sound worst and/or Pro Tools mix engine can not handle
all these calculations and will only make your mix/recording worst.

Myth - practically speaking.

The TDM buss has 24bit wide precision, which means that any signal presented to the bus has to be in that format.

Because properly designed plug-ins can process in double precision (or host 64bits for RTAS), the 24bit output can be the only source of reduction of precision and increased noise.

However a 24bit signal with dither is capable of 144dB of signal to noise ratio.

And since noise from multiple sources adds up by 3dB for every doubling of the number of sources (20Log(number of sources^0.5)) a very large number of 24bit sources are required to impact significantly on the signal to noise ratio of your mix.

For instance the signal to noise ratio of 256 24bit sources added together is

144 - 24 = 120dB

This is still around 30dB better than your output media.

So there's not much to worry about.

Quote:

8. Is This Truth Or Myth? -
It is better to stay totally ITB then to use a summing mixer for width/depth/headroom. Minus color of course.

Theoretically truth. One thing a computer can do is add stuff up! Analogue struggles with it.

Also having to convert into analogue and back into digital again is obviously a big potential source of error and signal quality loss.

Quote:

9. Is This Truth Or Myth? -
Even with today's cpu's, there is no way that we could exceed the sound of old reverb units. (Lexicon 480L/PCM90 etc)

Obviously this is a myth. The only thing that determines what reverb does is processing. The average home PC has many times the processing power of one of these legacy units.

The only thing preventing a plug-in being identical to these (if that is really desired) is that no one has actually bothered.

It is pretty obviously possible to exceed the performance of these units on any home PC.

Quote:

10. Is This Truth Or Myth? -
64bit floating point is better than 48bit fixed point. In fact, we should go 72bit to make it more analogy.

Myth practically speaking. A 48bit fixed point system is largely equivalent to a 64bit float system in real signal accuracy over the range of -1 to +1 - because the 64bit float system has a 52bit mantissa and a 11bit exponent.

The only case where this might affect things is in recursive processes (like filters etc.) where a floating system might benefit from the exponent.

However with 288dB of real dynamic range in a 48 bit system (float or otherwise) it's not an issue at all.

A 48bit fixed point system is equivalent to a 64bit float system in real signal accuracy over the range of -1 to +1 - because the 64bit float system has a 48bit mantissa and a 24bit exponent.

Flamewars is nothing for me but a bit of nitpicking never hurt anybody,64 bit float has 52 bit mantissa plus a sign bit. And since it's normalized you have 54 bits in total,no ? Floating point arithmetic does my head in.

Flamewars is nothing for me but a bit of nitpicking never hurt anybody,64 bit float has 52 bit mantissa plus a sign bit. And since it's normalized you have 54 bits in total,no ? Floating point arithmetic does my head in.

Yes sorry you are right it has

52bit mantissa 1 sign bit and 11 bit exponent

Excuse my addled brain and hasty typing - I was just thinking in terms of 32bit sorry. It doesn't change the fact that 48bit fixed is perfectly adequate for 24bit audio dynamic range.

3. Is This Truth Or Myth? -
Digital system's hardware only achieve 20hz - 20khz, there is no reason to have a plugin @ 10hz - 40khz? We can't hear and/or feel the difference.

Truth - if everything is done correcly.

Digital hardware even at base sampling rate can provide 0Hz - 20KHz.

I personally have no evidence (however much I've tried) that anyone can hear directly beyond that range.

Therefore it seems there is nothing much to be gained from producing supersonic freqs we can't hear that serve only to cause errors in repro systems.

Hi, Paul. Have to disagree there as some digital systems exceed the 20KHz threshold. As for supersonic freqs I can't dispute there. However if the limit is set at 20KHz as you know there are filters applied to set that limit that will affect amplitude and possibly frequency shifts near bandwidth limit.
Much respect...

How do you feel about the comparison of 32 bit floating-point vs. 64 bit? In the -1 to +1 range 32 bit FP has at least 24 bit accuracy (wasn't it 25 bit even because of one implicit bit?) which means 16,777,216 steps. Isn't this enough for pretty much any number crunching?

Hi, Paul. Have to disagree there as some digital systems exceed the 20KHz threshold. As for supersonic freqs I can't dispute there. However if the limit is set at 20KHz as you know there are filters applied to set that limit that will affect amplitude and possibly frequency shifts near bandwidth limit.
Much respect...

If I take this as a personal opinion question, yes we can sample higher and can obtain a greater bandwidth from doing so.

But in all the tests I have ever done on a system that was properly designed, I could find no way I could produce any signal (however complex, strange and deliberately difficult) that was changed in its sound by the 20KHz band limit on its own.

There may indeed be designs (which include filters) that do not function as well as they might, that may improve by sampling at higher rates, of course there must be.

But the question was about whether the band limit in itself was a fundamental cause of reduced sound quality.

From my own perspective, research and opinion I must be honest and say - no..

It does not have to be a limitation in sound quality.

This is where I start repeating answers I have made in many many threads over the last decade - and I know the answers will be as controversial as ever, because they seem counter-intuitive - nothing changes.. It's only a question of time before someone takes offense :-(

How do you feel about the comparison of 32 bit floating-point vs. 64 bit? In the -1 to +1 range 32 bit FP has at least 24 bit accuracy (wasn't it 25 bit even because of one implicit bit?) which means 16,777,216 steps. Isn't this enough for pretty much any number crunching?

32 bit float is definitely good enough to transfer music and store it - it's obviously at least as good as 24bt fixed point. But it's often not enough precision for processing within plug-ins.

Many common processes accentuate accuracy limitations, filters and EQ are one example. So therefore higher precision needs to be used in some cases to preserve the signal to noise ratio or provide the required degree of coefficient accuracy for the process.

It should also be noted that higher sampling rates increase requirement for precision if quality is to be preserved at higher rates, so this put yet more pressure on for double precision processing.

Now, I'm surprised this hasn't been asked in this thread already, or maybe I missed it somewhere... I, and many are already aware that the answer is yes, but just to hopefully hear from someone that's been working in audio DSP for a while...

Is This Truth Or Myth?
-Do all DAW's sound the same, except maybe for ones that are marketed to have extra "mojo" (Mixbus, Record)?

Many common processes accentuate accuracy limitations, filters and EQ are one example. So therefore higher precision needs to be used in some cases to preserve the signal to noise ratio or provide the required degree of coefficient accuracy for the process.

Does this happen in practice and do we know an example 32 bit plugin that adds noise due to precision boundaries?

With S/N ratio starting at -144 dB and over 16 mio quantization steps (over 33 mio if you consider the 25th implicit bit) I find it hard to imagine that even a chain of filters and EQ would push the final S/N ratio anywhere close to -100 dB (unless a plugin deliberately adds harmonics). Do I lack imagination or just the mathematical background?

But let's stick to the premise that 64 bit precision is better for filter and EQ plugins to be used *internally*. How much of an impact (precision loss) would the final *output* rounding to a 32 bit summing engine have? One reason why I ponder on the 32 vs. 64 bit question is because Jack and Port Audio still "only" offer 32 bit internally (for both transport *and* mixing).

Some current 64 bit implementations seem to do it the the other way around (Logic, Live), they output 32 bit plugins to a 64 bit summing path. Isn't that rather useless, especially with engines like Logic's that don't even offer 32 bit precision gain-stage controls/faders?

Quote:

It should also be noted that higher sampling rates increase requirement for precision if quality is to be preserved at higher rates, so this put yet more pressure on for double precision processing.

Why is that so?

Quote:

Luckily for host native processing this is not a costly thing to do.

While this is true for the CPU the main "cost" seems to be the coding effort.

Does this happen in practice and do we know an example 32 bit plugin that adds noise due to precision boundaries?

With S/N ratio starting at -144 dB and over 16 mio quantization steps (over 33 mio if you consider the 25th implicit bit) I find it hard to imagine that even a chain of filters and EQ would push the final S/N ratio anywhere close to -100 dB (unless a plugin deliberately adds harmonics). Do I lack imagination or just the mathematical background?

It has to do with feedback. The values gets smaller and smaller but because there is feedback there is lots of them so they can have a big impact on sound.
It's kinda hard to describe but it was a revelation for me when i first got it.

True for common processing that acts in the frequency domain like EQ etc..

Quote:

Originally Posted by Timur

Why is that so?

It's complex, but the simplest and most basic way of looking at it is that in the digital domain we only have delay as a frequency active element (there are no physical energy storing devices like capacitors and inductors).

So filters and EQs must act on the value differences between successive samples to function.

As the sampling rate increases the difference in values between samples at any particular signal frequency reduces, because their delay gets less, but the signal freq stays constant..

This means that to maintain signal performance the required precision of the numbers produced within the EQ algorithms must increase - simply because the differences are smaller relative to each other.

As a philosophical illustration, at an infinite sampling rate no frequency action would be possible, because there would never be any difference at all between samples

Where these values are used in feedback arrangements (like 2nd order sections in most EQs) the effect of these errors can be boosted to fairly high levels - especially when the EQ is set to low frequency action (where the difference between samples is least).

Insufficient precision in EQs can produce various results, from noise, low level distortion and limit cycles (birdies and tones) - all the way up to changing responses at low level or even simply failing to EQ at low levels at all!! I have seen several examples of this kind of behaviour in EQs in the early days - but luckily these are mostly consigned to history, with higher precision processing.

In the 1970's and 1980's many rearrangements of these algorithms were proposed to help this problem, each with it's own trade off. But as you would expect, you can't get something for nothing, so whatever apparent advantage a particular arrangement had, something else suffered instead. However some arrangements were more efficient with certain processor architectures etc.

Considering how people have been using 32 bit engines and still consider plugins to sound "better" at double-speed it seems that other factors than bit precision contribute much more to the *audible* result (i.e. anti-aliasing filter design)!?

Considering how people have been using 32 bit engines and still consider plugins to sound "better" at double-speed it seems that other factors than bit precision contribute much more to the *audible* result (i.e. anti-aliasing filter design)!?

I don't see how it's a contradiction. Specific implementations are being compared to theory. Theory only states what's possible to accomplish.

All implementations trade off a variety of factors. It's what engineering is all about. If one could simply look a few formulas up in a textbook and design a pretty graphical interface for them, we'd all just be using student projects from two decades ago.

Considering how people have been using 32 bit engines and still consider plugins to sound "better" at double-speed it seems that other factors than bit precision contribute much more to the *audible* result (i.e. anti-aliasing filter design)!?

Processing sounds different at different sampling rates, despite bit precision?
Truth or myth?

Yes they can - truth.

But it depends on the process itself and the quality of the design.

For instance an EQ with a cramped response towards the HF end will have less of a cramp if run at higher rates. One that does not cramp in the first place will be virtually unchanged.

EQ that oversamples inside to lose cramping (rather than using direct coefficient solutions) may perform better at higher rates, simply because the Plug-ins own internal downsampling filtering has to work less.

Compressors that have some kinds of non-linearity (either deliberately or by bad design) will alias somewhat less at higher sampling rates. (BTW remembering that aliassing is sometime deliberate to soften HF sounds).

ADCs and DACs with insufficient filtering (or phase compensation) may produce less artefacts at higher rates. Those that are well designed will not have those artefacts in the first place. And so on.

But the point here is that if a design is well thought out and designed carefully there is no reason why forcing the rest of the system to sample beyond baseband (and doubling the processing load) should change the sound of that application..

That's interesting. Could you go into that briefly or point me in the direction of another source of information? thanks thumbsup

Digital artefacts (sometimes considered errors) can be used for artistic effect?
Truth or Myth?

Truth.

Artefacts of digital system can be used for effect, in much the same way as analogue systems have been used for decades before.

The effects are of course different, but that should not stop engineers searching them out and using them artistically if appropriate. As has always been the case, this is often the role of a good sound engineer.

And of course designers of artist digital effects may employ digital artefacts in the intended design of their products, if they are appropriate for the sound they are trying to generate.

One of the biggest misperceptions in the digital arena is the notion of 'exactness' somehow being the only goal and justification for an artistic product. Whereas precision and exactness are a good things for, mixers, converters, monitoring and other reference equipment, needed in the signal chain, artistic applications require a more lateral view and should be based only on the intended sound character of the product - not some religious notion of correctness.

How the sound is generated should be secondary to the effect and character the designer is trying to create.